Digital data networks have become a ubiquitous part of business, commerce, and personal life throughout the United States and the world. The public Internet and private local and wide area networks (LANs and WANs) have become increasingly important backbones of data communication and transmission. Email, file access and sharing, and services access and sharing are but a few of the many data communication service and applications provided by such networks. Recently, next generation data communication applications such as Voice over IP (VoIP) and real-time interactive multi-media have also begun to emerge.
The Internet and nearly all digital data networks connected to it today adhere to substantially the same addressing and routing protocols specified in a “network layer” or “layer 3.” According to these protocols, each node in the network has a unique address, called the Internet Protocol (IP) address. To communicate digital data over the network or between networks, a sending or source node subdivides the data to be transmitted into “packets.” The packets include the data to be transmitted, the IP addresses of the source node and the intended destination node, and other information specified by the protocol. A single communication of data may require multiple packets to be created and transmitted depending on the amount of data being communicated and other well-known factors.
This approach introduces certain time considerations into the data communications process. Such time considerations arise for a number of reasons, including delays in the arrival of packets (latency) and delays due to the reconstruction of the signal because of variable inter-packet arrival time (packet jitter). For example, packets may be delayed in arrival if a specified or selected transmission route is interrupted due to problems (congestion) at an intermediary node. In such cases, further transmission may await resolution of the congestion at the intermediary node, which may result in even further delay. At the destination node, a certain amount of overhead is involved in processing packets in order to reconstruct their original sequence. Such overhead may increase substantially when a particular data communication involves a large number of packets, for example, or when the destination node is experiencing heavy processor loads due to other factors. In addition, it is possible for packets to be lost en route and to never reach the intended recipient node (packet loss). Further, since links are not dedicated and resources are shared, there are no guarantees of bandwidth availability.
VoIP provides real-time, interactive end-to-end voice communications over IP digital data networks using standard telephony signaling and control protocols. In VoIP, voice signals are converted to digital format, packetized, transmitted, and routed over the IP network from a source node to a destination node using the commonly used Internet protocols. At the destination, the packets are reassembled, and the voice signals reconstructed for play back. In VoIP, packet latency manifests itself as delay between the time one party to a conversation speaks and another party to the conversation hears what the speaker said. Delays that exceed a threshold and interfere with the ability to converse without substantial confusion are unacceptable. It has been demonstrated that one way packet latency in the range of 0 ms to about 150 msec results in excellent to good communication quality, whereas latency above about 150 msec results in poor to unacceptable quality.
Packets lost during transmission also adversely impact the quality of VoIP communications. It has been demonstrated that speech becomes unintelligible if voice packets comprising more than about 60 ms of digitized speech data are lost. Packets can be lost in transmission for three reasons: (1) if the electrical signal suffers from an electromagnetic disturbance, thus causing an error in one or more bits or (2) if the queue it is waiting in for transmission at some intermediary router along the path overflows due to congestion, thus causing packet dropping or (3) because of some configuration errors that cause packet transmission collisions (i.e., two or more electrical transmission signals overlapping and jamming each other). Because VoIP is a real-time interactive data communications application the current Internet protocols that provide for retransmission are of little help in this instance since late packets may become outdated, i.e., useless, when the packets finally arrive.
Packet jitter also substantially affects the quality of VoIP communications. In VoIP, packet jitter may result in the inability to reassemble all packets within time limits necessary to meet minimum acceptable latency requirements. As a consequence, sound quality can suffer due to the absence of some packets in the reassembly process, i.e., loss of some voice data or excessive packet delay. It has been determined that to achieve acceptable voice quality, voice packet inter-arrival times (i.e., jitter) generally must be limited to about 50–75 msec. Within this range, data buffering can be used to smooth out jitter problems without substantially affecting the overall quality of the voice communications.
Additionally, the current Internet addressing and routing protocols and approaches for fixed node data networks are incapable of supporting the dynamically changing addressing and routing situations that arise in recently proposed wireless, mobile-access digital data networks. The International Telecommunication Union (ITU) of the Internet Society, the recognized authority for worldwide data network standards, has recently published its International Mobile Communications-2000 (IMT-2000) standards. These standards propose so-called third generation (3G) and beyond (i.e., 3.5G, 4G, etc.) data networks that include extensive mobile Internet access by wireless, mobile node devices including cellular phones, personal digital assistants (PDA's), handheld computers, and the like. (See http://www.itu.int).
Unlike previous wireless networks, the proposed fourth generation and beyond networks are entirely IP based, i.e., all data is communicated in digital form via standard Internet addressing and routing protocols from end to end. However, unlike current fixed node networks, in the proposed third generation and beyond wireless, mobile access networks, wireless mobile nodes are free to move about within the network while remaining connected to the network and engaging in data communications with other fixed or mobile network nodes.
Among other things, such networks must therefore provide facilities for dynamic rerouting of data packets between the communicating nodes. The current Internet addressing and routing protocols and schemes, which are based on fixed IP addresses and fixed node relationships, do not provide such facilities. Similarly, current fixed node Internet protocols are not sufficient for wireless LAN usage.
Real-time applications such as VoIP usually have strict requirements over the end-to-end QoS from the mobile node all the way to the correspondent node. It is therefore necessary to ensure QoS requirements are met from source to destination and vice-versa, not just on portions of the path. The existing handoff trigger mechanisms in wireless communication systems are all essentially based on measurements of layer 2 QoS parameters over the wireless link only (BER—bit error rate, SNR—signal-to-noise ratio, etc.). Unfortunately, this measurement does not cover the path to the correspondent node in its entirety.
With circuit-switched core networks in current wireless communication systems, measurements of layer 2 QoS parameters over the wireless link only is sufficient to decide to which access point to handoff the wireless communication device. This is because the circuit-switched core network is robust and well provisioned to provide reliable and stable service. In that case, the wireless link is the only bottleneck in the end-to-end path and it is appropriate to base the handoff trigger on QoS measurements on that portion of the communication path.
As set forth generally above, next generation wireless communication systems will be based on an all-IP infrastructure, especially the core network. As such, these wireless communication systems will lose the robustness and reliability that was provided by circuit-switched networks thereby making the fixed hop of the end-to-end path from mobile to correspondent node a possible bottleneck, just like the wireless hop or link. To that end, using the existing layer 2 QoS measurements to determine handoff triggers, which do not estimate QoS on the fixed hop as well as the wireless hop, will lead to re-establishment of ongoing real-time sessions on paths where the end-to-end QoS requirements are not satisfied.
As such, a need exists in wireless communication systems that use an all-IP infrastructure for a handoff trigger that is based on the QoS experienced on the entire communication path between the wireless communication device and the correspondent node.